I guess you forward all calls via Asterisk. Yes: set canreinvite=yes (name was changed in newer Asterisk versions) in sip.conf for the peers and Asterisk will send reINVITEs after call setup to offload RTP.
regards Klaus Am 13.07.2011 07:43, schrieb MingHon: > Hi List, > > i would like to know is it possible to bypass the rtp traffic forwarding > to asterisk server? > > my kamailio and rtpproxy is on the same box and asterisk is on the other > box. > > can kamailio/rtpproxy handle the rtp traffic without forwarding to > asterisk box? > > thanks in advance. > > > -- > Regards, > > MingHon > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
